[..] |
ASTERISK-12000: Asterisk startup hangs if another alsa application is running |
ASTERISK-12001: Console-phone one-way audio option (chan_alsa.c/chan_oss.c) |
ASTERISK-12002: [patch] ExternalIVR will hang a channel attempting to read from a file descriptor that has somehow gone away |
ASTERISK-12003: added feature |
ASTERISK-12004: MeetMe conference freezes with Polycom |
ASTERISK-12005: [patch] Zap channels locking, staying in 'Rsrv' Mode and phone muted |
ASTERISK-12006: Using Addqueumember it's impossible to use the new way to define member |
ASTERISK-12007: The device state change thread gets locked (or waits) for 20 seconds |
ASTERISK-12008: rfc2833 DTMF problem |
ASTERISK-12009: Asterisk replies with VNAK (and wrong destination call) to AUTHREQ to its NEW (sometimes) when call bounced back. |
ASTERISK-12010: [patch] default for pridialplan in chan_zap national=>unknown |
ASTERISK-12011: CRBT using Asterisk PBX |
ASTERISK-12012: DTMF problem |
ASTERISK-12013: Giving remote command halts if asterisk process stops |
ASTERISK-12014: [patch] Variadic expansion in MONITOR_FILENAME |
ASTERISK-12015: app_chanspy group still broken in 1.4.19.2 |
ASTERISK-12016: Pressing # during the playback of vm-options within VoicemailMain causes the channel to lockup. |
ASTERISK-12017: Gtalk call from Empathy - no corresponding codecs |
ASTERISK-12018: [patch] Asterisk leaves zombie agi processes when running under linux 2.6 |
ASTERISK-12019: [patch] missed rwunlock in __ast_context_destroy(): deadlock |
ASTERISK-12020: [patch] Warn when a string of "(null)" is passed to ast_strlen_zero |
ASTERISK-12021: Whenever I issue reload, I get "No [csv] section in cdr.conf. Unregistering backend." |
ASTERISK-12022: Weird noise with Speex codec |
ASTERISK-12023: [patch] Autoservice loses DTMF digits |
ASTERISK-12024: [patch] DTMF issues on Zap |
ASTERISK-12025: [patch] Urgent folder specification incorrect when using IMAP storage |
ASTERISK-12026: [patch] Provide externnotify application with number of urgent messages |
ASTERISK-12027: [patch] String comparison of NULL Urgent flag value crashes Asterisk. |
ASTERISK-12028: High latency (satellite) hangup on internal calls |
ASTERISK-12029: [patch] MWI messages using MD message mutex and conditions |
ASTERISK-12030: Whenever I reload, asterisk stops logging to /var/log/asterisk/messages. |
ASTERISK-12031: Unable to overwrite SIP Header using SipAddHeader() |
ASTERISK-12032: [patch] fix module loading of chan_oss when you already got chan_console loaded (channel type conflict) |
ASTERISK-12033: Chan_skinny causing asterisk dumps core randomly |
ASTERISK-12034: some RTP packets sent to NAT IP instead of public IP; breaks built-in jitterbuffer on some phones |
ASTERISK-12035: Dynamic queue member does not survive reboot/restart when using aliases |
ASTERISK-12036: Device State stay stuck to "inuse" causing agent not to receive anymore call |
ASTERISK-12037: Setting same variable in call file multiple times only uses the value in first "Set" line |
ASTERISK-12038: [branch] revert ast_queue_hangup and create ast_queue_hangup_with_cause |
ASTERISK-12039: Astersik sometimes is crashing after sip reload |
ASTERISK-12040: It crashes on the command "reload" |
ASTERISK-12041: SIP Call completed elsewhere has 2 colons after Reason |
ASTERISK-12042: [patch] Transcoding breaks jitterbuffering |
ASTERISK-12043: Extension state change random delay |
ASTERISK-12044: Command Line - "core show channels" |
ASTERISK-12045: [patch] chan_h323 doesn't compile with pwlib 1.12.0 |
ASTERISK-12046: Queue timeout not overided by the timeout parameter on the Queue command |
ASTERISK-12047: Pressing "*" star by agent after receiving call, cause asterisk to hangup. |
ASTERISK-12048: [patch] enable chan_h323 to bind addr 0.0.0.0 |
ASTERISK-12049: asterisk crashing with mixmonitor thread |
ASTERISK-12050: [patch] rc.debian.asterisk ubuntu 8.04 hardy |
ASTERISK-12051: serious problems in VAD and CNG support |
ASTERISK-12052: [patch] Possible deadlock: rwlock can't be recursively locked (writelocked) |
ASTERISK-12053: [patch] Queue timeout terminates call attempt, causing partial ring |
ASTERISK-12054: [patch] chan_skinny doesn't respect callwaiting=no |
ASTERISK-12055: chan_skinny doesn't respect callwaiting=no |
ASTERISK-12056: sip reload doesn't add new peers |
ASTERISK-12057: [patch] Bad disposition on originated IAX2 calls |
ASTERISK-12058: Incorrect usage of errno can cause segfaults in ast_expr2.y |
ASTERISK-12059: [patch] missed result string length count |
ASTERISK-12060: [patch] ExternalIVR will never read from a socket |
ASTERISK-12061: Chan_skinny causing asterisk to crash |
ASTERISK-12062: Asterisk Allocates 3,1 GB of memory |
ASTERISK-12063: Zaptel channel detects hangup, but does not hangup bridged SIP channel |
ASTERISK-12064: Console displays strange "spade" characters at the start of every line |
ASTERISK-12065: ODBC connections are not being pooled or reused. |
ASTERISK-12066: [patch] Change the ExternalIVR() argument method, add 'I'nform event, add 'T'alk (answer channel) command, add options |
ASTERISK-12067: Violation of RFC 3261 |
ASTERISK-12068: Realtime/Rtcachefriends SIP Peer is not recognized as a Realtime / Cached peer, leading to rtautoclear not working. |
ASTERISK-12069: Dead air between answer and packet2packet bridge |
ASTERISK-12070: reload stops all action on CLI |
ASTERISK-12071: Crash on sip reload |
ASTERISK-12072: [patch] small patch to remove inefficient for loop to detect sequence numbers in iax2 |
ASTERISK-12073: MWI event mailbox and context strings in mwist struct destroyed at ast_taskprocessor_push |
ASTERISK-12074: [patch] SIP Protocol Violation when REFER rejected in sip_transfer (Cisco CCM, post answer), and Transfer application misclaims |
ASTERISK-12075: [patch] chan_ooh323 trunk missed some updates on 30 Jan 2007 |
ASTERISK-12076: [patch] warning about non-existent dialplan context is garbled, sometimes spurious |
ASTERISK-12077: [patch] chan_iax2 becomes unresponsive |
ASTERISK-12078: Configure function considers openh323 invalid while version 1.6 considers it valid |
ASTERISK-12079: When dial multiple SIP channels simutaneously, if one channel reply busy, the dial ended |
ASTERISK-12080: setting displayconnects=no still results in connection related messages (Parsing) |
ASTERISK-12081: [patch] Problems with Restore of fullcontact + ipaddr from RealtimeDB with rtcachefriends |
ASTERISK-12082: [patch] unterminated string in 1.4 compat mode |
ASTERISK-12083: Asterisk does not log all legs of call transfers and/or logs wrong information |
ASTERISK-12084: delayed RTP cause first sound to chop off |
ASTERISK-12085: ForkCDR application fails to set duration |
ASTERISK-12086: Missing notifications after call to non existing extension or failed global pickup attempt |
ASTERISK-12087: Ex-girlfriend-logic requires also most-specific extension matching |
ASTERISK-12088: [patch] Fix for vasprintf with printf format warning solaris |
ASTERISK-12089: Asterisk 1.4.19 give me the error "everyone is busy congested at this time" with TE405P card |
ASTERISK-12090: [patch] Separate directionality of log messages versus console commands |
ASTERISK-12091: Crash after 2 weeks of work |
ASTERISK-12092: [patch] Compatibilty wrapper for Macro to use GoSub |
ASTERISK-12093: core dumped while handling request bye |
ASTERISK-12094: Monitor fails to record call after second Dial(...) |
ASTERISK-12095: Asterisk should reply "200 OK" to an in-dialog INFO with empty body |
ASTERISK-12096: [patch] format_mp3.c update for change to struct ast_frame |
ASTERISK-12097: [patch] Multiparking support / AMI enhancement for Asterisk 1.4 |
ASTERISK-12098: Zap Calls get no audio |
ASTERISK-12099: [patch] OOH323 channel names shouldn't use CIDname |
ASTERISK-12100: AGI 100% CPU utilization (deadlock?) |
ASTERISK-12101: [branch] RT should adapt to existing columns on updates |
ASTERISK-12102: reINVITE --> Ignoring this INVITE request --> Hanging up call - no reply to our critical packet. |
ASTERISK-12103: Codec handling isn't working correctly |
ASTERISK-12104: [patch] multiparking, "WARNING[6006]: channel.c:2065 __ast_read: ..." fixed |
ASTERISK-12105: Awk script for ILBC not working throwing error |
ASTERISK-12106: [patch] failed to compile trunk - utils/check_expr: ast_bt_get_addresses() |
ASTERISK-12107: Asterisk stop responding to commands, stop processing calls in and out |
ASTERISK-12108: Autologoff is not working properly |
ASTERISK-12109: [patch] parking space number is announced to the parkee instead of the parker |
ASTERISK-12110: segfault in command port of chan_ooh323 |
ASTERISK-12111: [branch] when BRI span go down cannot make call |
ASTERISK-12112: [patch] Set fetchid to indicate bad connection id |
ASTERISK-12113: [patch] Corrupt Voicemail Notification E-Mails (with file attached.) |
ASTERISK-12114: [patch] Gosub invoked from AGI should do something |
ASTERISK-12115: [patch] chan_sip: build_contact() does not put alternate port setting in Contact header |
ASTERISK-12116: if queue_log file is FIFO asterisk do deadlock after "logger reload" |
ASTERISK-12117: chan_sip creates a new local tag (from-tag) for every register message |
ASTERISK-12118: build app_voicemail with IMAP_STORAGE .voicemessage forwording failed. |
ASTERISK-12119: Manager deadlock when loading a module |
ASTERISK-12120: [patch] Failure of resetting of a PRI B-Channel causes deadlock in process |
ASTERISK-12121: Asterisk crashes when switch => Realtime/@ is specified in the head of a context |
ASTERISK-12122: Multipile issues with chan_mobile |
ASTERISK-12123: 'core show channels' no longer shows active channels/calls |
ASTERISK-12124: dialed exten never fallback to 's' when it's not found, regression from #12479 |
ASTERISK-12125: [patch] Bogus <member> is still 'Not in Use' warnings for AgentLogin'ed agents. |
ASTERISK-12126: Trunk version of chan_sip significantly slower than 1.4.19.1 |
ASTERISK-12127: [patch] Agent doesn't propagate device state changes from underlying device. |
ASTERISK-12128: [patch]Log member adding and removing for realtime members |
ASTERISK-12129: Manager lock (?) |
ASTERISK-12130: Improve download links on Asterisk.org to allow wget/non-javascript browsers |
ASTERISK-12131: [patch] Pattern matching treats 'x' differently than 'X' |
ASTERISK-12132: [patch] Incorect duration values in INFO messages when emulation occurs |
ASTERISK-12133: [patch] smsq does not work in asterisk 1.6 |
ASTERISK-12134: [patch] logger.c may random crash asterisk on module unload for some OS |
ASTERISK-12135: Queue holdtime smoothing filter is not boxcar (FIR), but exponential average (IIR). |
ASTERISK-12136: no native bridging with SIP over TLS enabled |
ASTERISK-12137: [patch] Using AddQueueMember with the membername parameter causes minor issues |
ASTERISK-12138: app_dial should set DIALSTATUS to CONGESTION if unable to open TCP connection to SIP peer? |
ASTERISK-12139: chan_skinny softkey event crashing asterisk |
ASTERISK-12140: CALLERID is not logged correctly into cdrs |
ASTERISK-12141: Make all SQLite-related modules use Sqlite3 |
ASTERISK-12142: Gosub is broken in latest trunk |
ASTERISK-12143: func_odbc stopped working in current version |
ASTERISK-12144: Dropped calls with: Maximum retries exceeded on transmission |
ASTERISK-12145: app_fax not compile in last 1.6 svn |
ASTERISK-12146: iax2 deadlock asterisk |
ASTERISK-12147: [patch] CLI command "dnsmgr refresh" not registered |
ASTERISK-12148: Asterisk crashes when trying to make a call from Jabbin |
ASTERISK-12149: Asterisk SVN and chan_ooh323 |
ASTERISK-12150: fresh reboot of asterisk server, TCP sip peers get invite via UDP |
ASTERISK-12151: [patch] extenpatternmatchnew=yes breaks GotoIf |
ASTERISK-12152: Digit timeout giving negative values |
ASTERISK-12153: [patch] Add Pause/Unpause to cli interface |
ASTERISK-12154: [patch] one way only audio in MS-Office 2007 to asterisk calls |
ASTERISK-12155: [patch] ExternalIVR responds to the P command with a pipe delimited string, it should be comma delimited |
ASTERISK-12156: Deadlock in cahn_sip when using IMAP voicemail |
ASTERISK-12157: Transfer if use AgentLogin func not prop. work |
ASTERISK-12158: [patch] Superfluous AST_LIST_HEAD_DESTROY() for staticaly defined linked list in chan_agent and chan_local. |
ASTERISK-12159: status report in /etc/init.d/asterisk does not work |
ASTERISK-12160: Zap no reading PRI cause codes for PRI |
ASTERISK-12161: Realtime wrongly updates username to 's' |
ASTERISK-12162: [patch] changes use of usleep to nanosleep |
ASTERISK-12163: BRIDGEPEER variable not updated on attended transfer when codec translation is used. |
ASTERISK-12164: [patch] Always make chan_local report In Use when ringing another channel |
ASTERISK-12165: Attended transfer no audio |
ASTERISK-12166: If HANGUPCAUSE is 3 (No route to destination) why Asterisk replies a 503? |
ASTERISK-12167: SIP qualify option destroys the MWI subscription, causing no new MWI messages to be sent. |
ASTERISK-12168: Wrong peer selected when more than one to same host |
ASTERISK-12169: [patch] Allow alternate extensions to be assigned in users.conf |
ASTERISK-12170: UNIQUE ID field length too small |
ASTERISK-12171: Voicemail goes into an endless loop when wrong option is choosen under "Mail box options" |
ASTERISK-12172: Segfault on chan_sip with recording (when retransmitting?) |
ASTERISK-12173: Asterisk can not register as SIP client to Cisco BTS due to faulty CSeq header parsing |
ASTERISK-12174: [patch] "restart gracefully" can crash asterisk |
ASTERISK-12175: [patch] MeetMe() can't use 'X' and 's' options at the same time |
ASTERISK-12176: [patch] Chanspy audio is delayed or lost |
ASTERISK-12177: [Patch] update http.conf to reflect changes to default value for prefix |
ASTERISK-12178: It seems that bug 10467 is back |
ASTERISK-12179: [patch] Makes pbx_builtin_getvar_helper values thread safe |
ASTERISK-12180: [patch] Setting "announce-holdtime" to "once" does not mention the holdtime at all. |
ASTERISK-12181: mysql libs not found after installing the plugin into asterisk |
ASTERISK-12182: cdr_tds won't build against FreeTDS 0.82 |
ASTERISK-12183: unable to use switch in macro include file |
ASTERISK-12184: asterisk fails to load chan_dahdi.so |
ASTERISK-12185: use ast_getformatname_multiple instead of ast_getformatname for log message |
ASTERISK-12186: DTMF detection issue in GSM gateway |
ASTERISK-12187: [patch] Page(CHANNEL) doesn't bring 'CHANNEL' into the MeetMe conference. |
ASTERISK-12188: accessing the console using the "-rx" option causes a significant delay |
ASTERISK-12189: agents.conf: ackcall=always is not documented |
ASTERISK-12190: Asterisk fails to start |
ASTERISK-12191: Unparked caller has ability to transfer |
ASTERISK-12192: [patch] users.conf mailbox |
ASTERISK-12193: [patch] add an IAXCHANINFO function similar to SIPCHANINFO |
ASTERISK-12194: Hangup extension doesn't seem to work in ast-1.4.19.1 |
ASTERISK-12195: "queue show" shows old (cached) information, whilst "queue show QUEUENAME" shows current information when using realtime. |
ASTERISK-12196: realtime support and qualify (sip_poke_all_peers) after restart |
ASTERISK-12197: [patch] Included example schema not compatible with res_config_ldap.c |
ASTERISK-12198: [patch] incorrect state update in friend nodes (chan_sip.c) |
ASTERISK-12199: Asterisk gives up on registration after receiving 408 Timeout response once |
ASTERISK-12200: IAX2 crashes with many leftover threads. |
ASTERISK-12201: [patch] Don't send BYE for a dialog that already was terminated after blind transfer |
ASTERISK-12202: Bad quality audio when using Local channel with a call file |
ASTERISK-12203: chan_iax2 becomes unresponsive. Issue 0012717 back |
ASTERISK-12204: [patch] 'context' doesn't change when 'sip reload' issued when driven from realtime |
ASTERISK-12205: Jerry Geis's dialplan 'works in 1.4, but not in 1.6' problem |
ASTERISK-12206: chan_zap not dropping inbound ISDN call. |
ASTERISK-12207: Memory Leak in HTTP Manager Interface |
ASTERISK-12208: [patch] Asterisk -1.4.21 gives core dump |
ASTERISK-12209: Retreiving some voicemails from ODBC store causes core dump |
ASTERISK-12210: [patch] small bug introduced with fix for 12656 |
ASTERISK-12211: CallerID missing on IAX channels |
ASTERISK-12212: [patch] Support for CDR's and syslogd |
ASTERISK-12213: Navtel Interwatch 9500 Fails Interop Test with Asterisk |
ASTERISK-12214: Asterisk Fails SIP Performance Test with Navtel Interwatch 9500 Hardware Call Generator |
ASTERISK-12215: Asterisk Fails SIP Performance Test with Navtel Interwatch 9500 Hardware Call Generator |
ASTERISK-12216: Queue() uses cached information to determine whether queue is empty or not |
ASTERISK-12217: app fax deleted in main tree and don't added to addons tree |
ASTERISK-12218: audiohook.c always throws in debug: write factory XXX was pretty quick last time, waiting for them |
ASTERISK-12219: build_tools/strip_nonapi 1.4/1.6 may run strip with -N as last argument |
ASTERISK-12220: [patch] Queue is treated as empty if it isn't, but no agents meet the QUEUE_MIN_PENALTY and QUEUE_MAX_PENALTY criteria |
ASTERISK-12221: Asterisk should reply 480(instead of 404) when a called user is not registered |
ASTERISK-12222: crash asterisk.c compiled for i486sx with out fpu |
ASTERISK-12223: [patch] chan_misdn.c threadsafe patch |
ASTERISK-12224: [patch] Small patch to 1.4 for report origposition then event is Transfer |
ASTERISK-12225: notifyringing option comment in configs/sip.conf.sample |
ASTERISK-12226: Incorrect format of events send to the child's stdin - missing second coma. |
ASTERISK-12227: [patch] Fix comparison logic to determine urgent versus non-urgent. |
ASTERISK-12228: [patch] "Your message has been saved" played twice. |
ASTERISK-12229: [patch] IMAP attachments not handled properly. |
ASTERISK-12230: [patch] Forwarding of an IMAP message may set the message to Urgent inadvertently. |
ASTERISK-12231: Cannot transfer voicemail in IMAP |
ASTERISK-12232: [patch] IMAP forwarding does not go to the correct mailbox. |
ASTERISK-12233: Asterisk crashes sometimes, when using Mixmonitor |
ASTERISK-12234: typographic error page 78 of Asterik 2nd edition |
ASTERISK-12235: DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP |
ASTERISK-12236: Video RTP is not sended to originating SIP extension when using IAX2 to interconnect both servers |
ASTERISK-12237: REGRESSION: chan_iax2.c patch 122259 results in chan_iax2.c sending no PING or LAGRQs |
ASTERISK-12238: [patch] Missing ChannelType in PeerStatus manager event. |
ASTERISK-12239: Handful of problems with DIGIUM-MIB and ASTERISK-MIB |
ASTERISK-12240: NoCDR() is not working |
ASTERISK-12241: Wrong log level for one liner (ast_log LOG_WARNING) generates too much logging |
ASTERISK-12242: [patch] MusicOnHold start/stop event. |
ASTERISK-12243: ForkCDR() with 'v' option crashes * - only when 'DONT_OPTIMIZE' flags is disabled. |
ASTERISK-12244: [patch] proper dependencies on dahdi/tonezone |
ASTERISK-12245: An issue with the IAX2 channel allows anonymous connections to cause resource starvation |
ASTERISK-12246: DTMF not reproduced towards ZAP T1 Port after connection has arrive as SIP Trunk |
ASTERISK-12247: asterisk use 99% cpu when agi script uses $AGI->exec("Dial","Local/ext@extensions") |
ASTERISK-12248: Multiple incoming call to free agent |
ASTERISK-12249: It crashses every 4 minutes |
ASTERISK-12250: Deadlock |
ASTERISK-12251: Segmentation fault in "Session timer expired:" debug message |
ASTERISK-12252: [patch] Added CURL() Function Timeout Argument |
ASTERISK-12253: Asterisk 1.4.21 breaks realtime sip on 'sip reload' |
ASTERISK-12254: ap_queue does not call the application specified in monitor_exec when monitor_type=monitor |
ASTERISK-12255: ap_queue hangs up caller |
ASTERISK-12256: IAX2 channel gets stuck, causes CLI to get stuck, * won't restart |
ASTERISK-12257: Only the last agent in a queue get added |
ASTERISK-12258: Asterisk-1.4.21 becomes unresponsive |
ASTERISK-12259: [patch] Compile failure on tonezone_compat.h with zaptel 1.4 and asterisk 1.4 |
ASTERISK-12260: retrydial hangs up when using a silent soundfile as anouncement |
ASTERISK-12261: retrydial hangs up when using a silent soundfile as anouncemen |
ASTERISK-12262: [patch] T38 gateway for asterisk 1.4 |
ASTERISK-12263: [patch] Union rather than cast for outbuf (translate.h) |
ASTERISK-12264: [Patch] Add verbose option to astcli |
ASTERISK-12265: Segmentation fault in ast_atomic_fetchadd_int |
ASTERISK-12266: asterisk-1.6.0-beta9 crash, not sure why |
ASTERISK-12267: Mod event socket |
ASTERISK-12268: DTMF package's ssrc number wrong when partical bridge channel |
ASTERISK-12269: setting volgain results in tempfile errors and 0-length email attachments |
ASTERISK-12270: iax2 hangs with jitterbuffer or timestamps - after hang it does not accept calls |
ASTERISK-12271: Failing registrations to peer via DNS name breaks all other peers |
ASTERISK-12272: [patch] Dialog state NOTIFY for hold uses badly formed XML |
ASTERISK-12273: SIP Channel Reports Wrong CDR |
ASTERISK-12274: [patch] ResetCDR does not work on non-answered channel |
ASTERISK-12275: [patch] Column names causes |
ASTERISK-12276: config call back... |
ASTERISK-12277: [patch] Fix solaris build issue acl.c |
ASTERISK-12278: [patch] PacketCable NCS 1.0 Support for Docsis / Eurodocsis Networks |
ASTERISK-12279: [patch] Busy / Congestion responses are not sent reliably |
ASTERISK-12280: Asterisk can not insert CDRs into cd table due to incorrect column name quoting... |
ASTERISK-12281: [patch] Segfault at make_email_file |
ASTERISK-12282: suspected typo in main/rtp.c bridge_p2p_rtp_write() payload type check (can cause RFC2833 DTMF detection issues) |
ASTERISK-12283: [patch] Convert chan_skinny's open-coded linked lists to the list macros |
ASTERISK-12284: Call rejected with 403 when sending a call between two SIP gateways |
ASTERISK-12285: Queue members as SIP/XXXX do not update status correctly |
ASTERISK-12286: using usb audio on chan_alsa seg fault |
ASTERISK-12287: ast_merge_contexts_and_delete() does not merge switches and includes properly |
ASTERISK-12288: [patch] allow pbx_lua to be used without pbx_config loaded |
ASTERISK-12289: [patch] Change the "sip notify" CLI to allow for adding message body to the SIP NOTIFY message (Patch submission) |
ASTERISK-12290: [patch] chan_iax2 will create multiple sessions when receiving retransmitted NEW |
ASTERISK-12291: Asterisk Now browser compatibility. |
ASTERISK-12292: Local dial with moh causes crash |
ASTERISK-12293: "make samples" gets an error "/bin/sh: line 44: Set: command not found" |
ASTERISK-12294: chanspy - crashes Asterisk - still |
ASTERISK-12295: Agent Status and outgoing calls |
ASTERISK-12296: [patch] Fix New Call softkey handling and cleanup phone display after a transfer |
ASTERISK-12297: Chanspy crashes when spying on a channel |
ASTERISK-12298: The strategy roundrobin or rrmemory does not work, only ringall |
ASTERISK-12299: [patch] Don't override/duplicate optimization flags. |
ASTERISK-12300: Crash during make_email_file() when cidname is originally an empty string |
ASTERISK-12301: [patch] Using VoiceMail() with IMAP when cid_num or cid_name is blank always leads to segfault |
ASTERISK-12302: No clear license for sound files |
ASTERISK-12303: Asterisk lock - Becomes unresponsive |
ASTERISK-12304: [patch] app_queue does not handle Custom: state interfaces |
ASTERISK-12305: [patch] "Dialplan remove extension" Command cannot remove extension that matches specific CID |
ASTERISK-12306: App ices |
ASTERISK-12307: CDR(billsecs) is not set if the M() option is used and the dialed party answers but hangs up whilst the macro is running |
ASTERISK-12308: [patch] Retransmitted RFC 2833 RTP events do not increment the RTP sequence number |
ASTERISK-12309: [patch] meetme threadsafe janitor |
ASTERISK-12310: [patch] app_chanspy threadsafe janitor |
ASTERISK-12311: [patch] segfault in app_chanspy.cpp |
ASTERISK-12312: IMAP (apparently) causing system crash |
ASTERISK-12313: Incomming rfc2833 DTMF is not relayed via SIP INFO method |
ASTERISK-12314: Asterisk cli filename completion (completion_fn_2&3). Crash when asterisk could not open a directory. |
ASTERISK-12315: Queues not reporting agent status correctly |
ASTERISK-12316: Reopening this issue -----> 0012927: Asterisk-1.4.21 becomes unresponsive |
ASTERISK-12317: [patch] Spamming CLI / logs with 'Remote host can't match request BYE to call...' |
ASTERISK-12318: res_config_ldap crashes consistently with "Illegal Instruction" |
ASTERISK-12319: [patch] SIP channel cannot handle FLASH HOOK INFO message sent by SIP device |
ASTERISK-12320: [patch] Implementation of application/telephone-event for SIP INFO |
ASTERISK-12321: [patch] Remove the 500ms delay trying to receive all CLI command output. |
ASTERISK-12322: [patch] janitor task to substitute in-line sizeof() for array_len to calculate the length of arrays |
ASTERISK-12323: [patch] xml doc for xmldoc branch |
ASTERISK-12324: Recording speed too fast (running BRI B410P) |
ASTERISK-12325: [patch] bug in contrib/init.d/rc.redhat.asterisk startup script |
ASTERISK-12326: [patch] chan_sip ignores rport and does not reply to source IP:port |
ASTERISK-12327: [patch] dynamic thread identifiers may be duplicated |
ASTERISK-12328: [patch] find_idle_thread() uses spin wait |
ASTERISK-12329: File Revision in SVN |
ASTERISK-12330: [patch] unwrap_timestamp() is bogus for video frames |
ASTERISK-12331: [patch] ast_iax2_new() fails to account for a bad situation, deals with another incorrectly |
ASTERISK-12332: Unable to access voicemail option zero (0) in order to record greeting messages |
ASTERISK-12333: [patch] new CLI commands for handling dialplan variables and name standartization for the old ones |
ASTERISK-12334: [patch] spaces after newlines in logging |
ASTERISK-12335: [patch] new feature: iax2 encryption key rotation at 2-5 minute intervals. |
ASTERISK-12336: MusicOnHold playing silence |
ASTERISK-12337: [patch] better interoperability for app_fax |
ASTERISK-12338: [patch] better interoperability for app_fax |
ASTERISK-12339: CDR does not get written |
ASTERISK-12340: G722 negociation problem with Aastra 5xi phones |
ASTERISK-12341: Call blocked after 1 minute 45 seconde |
ASTERISK-12342: [patch] Zaptel/DAHDI channel name don't change after a hookflash event |
ASTERISK-12343: [patch] Add support to handle incoming out-of-dialog SIP NOTIFY requests for "message-summary" event package |
ASTERISK-12344: asterisk use 99% cpu |
ASTERISK-12345: [patch] Janitor project to add channel locks |
ASTERISK-12346: [patch] 183 response although progressinband=never |
ASTERISK-12347: ChanSpy multiple channels attached to one |
ASTERISK-12348: Caller gets dropped and engaged tone after two Zap channels bridged. |
ASTERISK-12349: Asterisk incorrectly requires an ACK of it's response to an OPTIONS request |
ASTERISK-12350: [patch] It is time to remove all ready deprecated CLI command 'show parkedcalls'. |
ASTERISK-12351: 1.4.21.1 crashes seg fault using console/dsp |
ASTERISK-12352: Asterisk crash while unloading pbx_ael.so |
ASTERISK-12353: [patch] Add missing state_interface on the dump_queue_members function |
ASTERISK-12354: No voice joining snom 190 through asterisk to cisco voice gateway occationally. |
ASTERISK-12355: Asterisk segfault at startup/ael reload |
ASTERISK-12356: new added function queue_transfer_fixup crashes asterisk |
ASTERISK-12357: Memory segmentation fault on T.38 pass through |
ASTERISK-12358: SIPPEER isn't working inside REGEX and IF |
ASTERISK-12359: Called Party's inband DTMF removed almost entirely with overlapdial and non-native bridging |
ASTERISK-12360: [patch] janitor project to use ARRAY_LEN instead of in-line sizeof() and division. |
ASTERISK-12361: rtptimeout and rtpholdtimeout can only be set globally for users |
ASTERISK-12362: [patch] Fix 'core show sysinfo' command, it is returning all 0s. |
ASTERISK-12363: [patch] ast_str api janitor |
ASTERISK-12364: Sound installer extracts with originating user/group ids. |
ASTERISK-12365: Setting toneduration in zapata.conf causes chan_zap not load any more |
ASTERISK-12366: [patch] crash can occur when ast_channel_free() tries to free a chan_iax2 tech_pvt |
ASTERISK-12367: random appears to be broken |
ASTERISK-12368: [patch] doc/tex/Makefile uses non-portable -i flag for sed |
ASTERISK-12369: Crash in iax2_destroy at chan_iax2.c:1309 |
ASTERISK-12370: [patch] add some const qualifiers |
ASTERISK-12371: [patch] sip_nat_settings - script to generate externip and localnet |
ASTERISK-12372: [patch] Free cfg structure on error. |
ASTERISK-12373: [patch] remove some erroneous trailing newlines from calls to astman_send_error() (and one call to astman_send_ack() ) |
ASTERISK-12374: [patch] fix a memory leak in an error handling path |
ASTERISK-12375: [patch] put 'static' at beginning of declarations |
ASTERISK-12376: [patch] OPTIONS response on default port. |
ASTERISK-12377: [feature request] Charge Number (ANI) is not delivered through SIP |
ASTERISK-12378: [patch] Memory leak while handling device state change. |
ASTERISK-12379: [PATCH] duplicate code for setting LSPI |
ASTERISK-12380: Re-Invite occurs eventhough the codecs are incompatible. |
ASTERISK-12381: Forwarded voicemail with prepended message replaces original voicemail |
ASTERISK-12382: menuselect compilation failure on Solaris 10 / gcc 3.4.3 |
ASTERISK-12383: [patch] in ast_cdr_setapp if (!app) app = NULL; if (!data) data = NULL; is redondant with following uses of S_OR |
ASTERISK-12384: [patch] channel variable SENDTEXTSTATUS is not being set if the channel doesn't support send_text |
ASTERISK-12385: [patch] Use ast_free() instead of free() for freeing memory in asterisk applications. |
ASTERISK-12386: [patch] Enable failover in func_odbc |
ASTERISK-12387: Once enabled the recording/monitoring will not disable |
ASTERISK-12388: [patch] logrotate script for contrib/scripts |
ASTERISK-12389: 99% cpu with 'logger rotate' |
ASTERISK-12390: Hangug MFCr2 Brazil-Telefonica Sao Paulo |
ASTERISK-12391: core show locks does not show all locks |
ASTERISK-12392: multiple "infinite loop" messages being output by AEL compiler. |
ASTERISK-12393: IAX2 username is case-insensitive in registration, but case sensitive in dialing |
ASTERISK-12394: [patch] don't dereference null in an error handling path |
ASTERISK-12395: ./configure --prefix=/usr/local fails to set sysconfdir properly. |
ASTERISK-12396: Call Fails to go to extension using WaitExten |
ASTERISK-12397: GMail Support |
ASTERISK-12398: [patch] avoid a leak in an error handling path |
ASTERISK-12399: [patch] several fixes in res_config_sqlite |
ASTERISK-12400: [patch] p->ackcall not reset when agent logs out |
ASTERISK-12401: [patch] avoid some leaks in error handling paths |
ASTERISK-12402: [patch] fix content of CHANGES file for sip.conf |
ASTERISK-12403: [patch] Memory leak if no feature group or feature specified in main/features.c/register_group_feature() |
ASTERISK-12404: [patch] The duration of each digit is also accepted to be passed as a param, and wasn't documented. |
ASTERISK-12405: [patch] Add 'iax2 set debug peer <peer>' |
ASTERISK-12406: [patch] Fix bogus uses of ASTERISK_FILE_VERSION SVN revision keyword string |
ASTERISK-12407: [patch] If context wasn't found in extensions.conf - don't give up, look in RT. |
ASTERISK-12408: [patch] Warning on unknown config directive #something |
ASTERISK-12409: ERROR[24649]: app_dial.c:1577 dial_exec_full: Could not stop autoservice on calling channel |
ASTERISK-12410: [patch] correct the use of errormsg according to the API and implementation of SQLite. |
ASTERISK-12411: [patch] fix comments |
ASTERISK-12412: [patch] /usr/sbin/safe_asterisk: 31: Syntax error: "(" unexpected |
ASTERISK-12413: [patch] Can't build RPM with 1.4.21.1 (few errors & fix) |
ASTERISK-12414: [feature request] Cannot bind selectively to multiple IP addresses |
ASTERISK-12415: could NOT get the channel lock |
ASTERISK-12416: Asterisk 1.4.21.1 segfaults many times daily using mixmonitor |
ASTERISK-12417: [patch] multiple 'transport' on peer doesn't work, tcp port still open |
ASTERISK-12418: Duplicated Digit 9 DTMF Relay/ Origination from O2 Germany |
ASTERISK-12419: During call transfer INVITE is sent twice without waiting for reply. |
ASTERISK-12420: Sip to Sip dial and rtsavesysname not working in latestsvn |
ASTERISK-12421: [patch] segmentation fault with chan_h323 ast_rtp_new_source (rtp=0x0) |
ASTERISK-12422: [patch] Search channel by name prefix fails if a previous channel was defined in ast_channel_find_locked. |
ASTERISK-12423: SIP/2.0 400 SIP Parser Error : Missing '@', line 3, column 26 |
ASTERISK-12424: [patch] Add CallerIDName and CallerIDNum on MeetmeJoin event |
ASTERISK-12425: [patch] basic documentation missing for a few options |
ASTERISK-12426: [patch] Fix some issues related to CLI command 'iax2 set debug' and also add peer autocomplete for 'iax2 set debug peer' |
ASTERISK-12427: rtcachefriends=no causing endless loop resulting in unusable asterisk |
ASTERISK-12428: Orphaned SIP Channel when accessing voice mail application |
ASTERISK-12429: [patch] dummy-select - a simplified menuselect replacement |
ASTERISK-12430: [patch] Some fixes to autocompletion in some commands. |
ASTERISK-12431: [patch] Make app_image work like app_sendtext, app_url, etc. |
ASTERISK-12432: [patch] SIPPEER(peername, chanvar[varname]) not working as expected. |
ASTERISK-12433: [patch] sip peer qualified failed, asterisk lock. |
ASTERISK-12434: [patch] IAXPEER() parameter separator must be comma (,) and not pipe (|), also remove the already deprecated use of ':' |
ASTERISK-12435: [patch] Update func_dialgroup to store in astdb |
ASTERISK-12436: Parking with hints enabled crashes Asterisk |
ASTERISK-12437: [patch] Setting up a HANGUPCAUSETEXT variable for SIP channel |
ASTERISK-12438: [patch] Save to alternate folder throws errors due to incorrect IMAP mailbox specification in Cyrus IMAP |
ASTERISK-12439: [patch] Added format_png to asterisk. |
ASTERISK-12440: NUEVAS VOCES DE ASTERISK EN ESPAÑOL CON ACENTO NEUTRO!!!!!! |
ASTERISK-12441: [patch] asterisk.logrotate missing '/var/log/asterisk/messages' and '/var/log/asterisk/debug' is duplicated |
ASTERISK-12442: [patch] add logrotate support to Makefile |
ASTERISK-12443: [patch] Sending silence while recording |
ASTERISK-12444: Logger doesn't show correct PID |
ASTERISK-12445: Queues: retry setting has no effect |
ASTERISK-12446: [patch] extending the support of ATTENDED_TRANSFER_COMPLETE_SOUND for more channel types |
ASTERISK-12447: [patch] set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path |
ASTERISK-12448: Asterisk crash when compile with low_memory |
ASTERISK-12449: [patch] Hebrew support for app_voicemail |
ASTERISK-12450: Local channel have emty account code in CDR |
ASTERISK-12451: Builtin application NoOp is empty |
ASTERISK-12452: No application 'DigitTimeout' for extension |
ASTERISK-12453: Memory leak in iax2-parser.c |
ASTERISK-12454: Commit 128812 breaks app_voicemail on CentOS 5 |
ASTERISK-12455: [patch] asterisk.ldif version not updated |
ASTERISK-12456: Asterisk 1.4 doesn't save file name of an agent recording in cdr userfield |
ASTERISK-12457: [patch] Add various app_dial features to app_bridge |
ASTERISK-12458: call parking in one step + retrieving the parkingslot |
ASTERISK-12459: call parking in one step + retrieving the parkingslot |
ASTERISK-12460: 1.6beta 4 crashes with ChanSpy |
ASTERISK-12461: On MacOSX 10.5.4 asterisk 1.4.21.2 dies on a poll.c error |
ASTERISK-12462: [patch] 'module unload chan_iax2.so' causes errors on cli, however module unloads |
ASTERISK-12463: [patch] Manager event 'Hangup' with no CallerIDNum and CallerIDName. |
ASTERISK-12464: [branch] Asterisk blocked when 2 or more users leave a meetme when announce user is on |
ASTERISK-12465: [patch] Allow to set the current username and groupname in the asterisk CLI prompt |
ASTERISK-12466: AST_CONFIG() doesn't handle correctly, if variable isn't found |
ASTERISK-12467: Server looses Registration Status / Client does not re-register |
ASTERISK-12468: Can not substring a variable using variables |
ASTERISK-12469: Irrelevant Notice |
ASTERISK-12470: [patch] Called Party id isn't handled correctly |
ASTERISK-12471: [patch] Rename rrmemory strategy to roundrobin |
ASTERISK-12472: [patch] Missed Calls is cleared from phone status line |
ASTERISK-12473: crash when using asterisk db put |
ASTERISK-12474: The chan_iax2 does not detect or send hangup to softphone |
ASTERISK-12475: Attended transfers call is lost |
ASTERISK-12476: Queue timeout doesn't work when periodic-announce is set. |
ASTERISK-12477: vmail.cgi does not look in users.conf -- misses users defined through AsteriskGUI |
ASTERISK-12478: "State:Idle" when placing outgoing calls |
ASTERISK-12479: Injection data sond in Channel user |
ASTERISK-12480: PRI span is locking up frequently |
ASTERISK-12481: Incoming ISDN call gives segfault in manager_isdn_handler |
ASTERISK-12482: Voicemail left in wrong mailbox if multiple users have the same mailbox # but are in a diferent context |
ASTERISK-12483: chan_sip fails to parse reason parameter in Diversion: header |
ASTERISK-12484: QueueMember event for QueueStatus action in AMI does not handle memberfilter correctly. |
ASTERISK-12485: STAT() inside IF() doesn't work |
ASTERISK-12486: Asterisk sends a [slightly] different branch id back on CANCEL than on INVITE when call times out |
ASTERISK-12487: Channel var AGISTATUS is not set for many AGI/DeadAGI failure conditions |
ASTERISK-12488: Bridging channels too early when Macro is used to control callee leg |
ASTERISK-12489: When I send a call it exits memory |
ASTERISK-12490: When I try to load asterisk it crashes |
ASTERISK-12491: [patch] [branch] [appdocsxml] Some fixes, and trying to start an entry point for patches to this branch |
ASTERISK-12492: Crash during internal proccessing of state change |
ASTERISK-12493: DTMF RFC2833 via SIP is not working |
ASTERISK-12494: Asterisk production environment crashes several times during day |
ASTERISK-12495: [patch] Fix a TeX issue on two documentation files. |
ASTERISK-12496: dundi precache not working |
ASTERISK-12497: [patch] DAHDI mwi thread race conditions, sometimes crash or output wierd noises on the call |
ASTERISK-12498: Asterisk crashes or locks up when leaving imap messages |
ASTERISK-12499: Several deadlocks in skinny |
ASTERISK-12500: [patch] Incorrect ANSWERTIME when using M option |
ASTERISK-12501: asterisk stops sending qualify |
ASTERISK-12502: [patch] when insecure variable can't be retrieved, asterisk crashes |
ASTERISK-12503: possible missing unlock |
ASTERISK-12504: [patch] Calls in high-weighted queue block low-weighted |
ASTERISK-12505: ASTSBINDIR isn't defined in rc.mandrake.asterisk |
ASTERISK-12506: Asterisk commands "moh reload" or "reload res_musiconhold.so" causes MOH not to work properly |
ASTERISK-12507: [patch] Generate a version when asterisk is built in a git repository |
ASTERISK-12508: chan_skinny crashes on registration of non-configured device |
ASTERISK-12509: CLI is freeing sometimes null values |
ASTERISK-12510: zt_setoption segfault in hangup in pri_dchannel |
ASTERISK-12511: Asterisk does not send voicemail alerts when it is being executed as a non-root user |
ASTERISK-12512: When calling a queue, after a few loops over a madplay'ed file, the MOH ceases to be played |
ASTERISK-12513: [patch] Free ast_config structure on memory allocation error. |
ASTERISK-12514: [patch] When unloading app_voicemail, it is not freeing up all the allocated memory. |
ASTERISK-12515: [patch] iax2-provision is not freeing iax_templates structure. |
ASTERISK-12516: Asterisk reports dialstatus as "CONGESTION" when received a 480 "Temporarily unavailable" |
ASTERISK-12517: [patch] Incorrect behavior when moving messages to/from New/Old folders |
ASTERISK-12518: Memory leak in Asterisk 1.4 and Trunk |
ASTERISK-12519: doesn't set ~~EXTEN~~ on the rigth place when a switch statement has been found |
ASTERISK-12520: Did a code review and found a few issues |
ASTERISK-12521: False state in core show hints |
ASTERISK-12522: [patch] Queue timeouts not always working correctly. |
ASTERISK-12523: [patch] Execute Playtones(Busy) from AGI and Asterisk crash |
ASTERISK-12524: [patch] [sound] Ability to force forwarding WITH comment |
ASTERISK-12525: SIP registration does never work, in case DNS not available during firstregistration |
ASTERISK-12526: [patch] Set(SIP_CODEC=xxxx) only applies to first inbound leg of call |
ASTERISK-12527: changes to menuselect-tree but apprently not in Changelog. This change causes my build to fail |
ASTERISK-12528: Stops replying to some registration requests after restarting the process or rebooting server |
ASTERISK-12529: crash related to ast_rtp_new_source |
ASTERISK-12530: Channel hangup on iax transfer when dialplan in DB |
ASTERISK-12531: AEL does not translate quoted strings correctly in 1.6 |
ASTERISK-12532: No hangup after attended transfer |
ASTERISK-12533: [branch] endbeforehexten=yes is useless now |
ASTERISK-12534: [patch] set FAXMODE variable to let dialplan know what fax transport was used |
ASTERISK-12535: Inband dtmf not working in voicemailmain |
ASTERISK-12536: bad contact port keeps carriers from taking calls |
ASTERISK-12537: crash if hangup during wait to read |
ASTERISK-12538: crash if hangup during sip_read |
ASTERISK-12539: Registering asterisk to other port apart from by default 5060 port |
ASTERISK-12540: Problem of call-back |
ASTERISK-12541: [Solaris] Build fails with new files added in asterisk 1.2.28.1 and above |
ASTERISK-12542: Problem of Voicemail |
ASTERISK-12543: Forwarding a voicemail message with no envelope fails |
ASTERISK-12544: updateconfig adding lots of unnecessary new lines randomly inbetween lines and contexts |
ASTERISK-12545: NoCDR does not work as expected in latest SVN |
ASTERISK-12546: chan_vpb.cc:401: sorry, unimplemented: non-trivial designated initializers not supported |
ASTERISK-12547: RFC2833 mangled from Sonus when RTP stream passes through asterisk rather than being reinvited |
ASTERISK-12548: parking an outbound calls loses GROUP() channel count information |
ASTERISK-12549: parking an outbound calls loses GROUP() channel count information |
ASTERISK-12550: how to change the default encoding format from pcm into alaw |
ASTERISK-12551: how to change the default encoding format from pcm into alaw |
ASTERISK-12552: [PATCH] ODBC database handles allocated but never released |
ASTERISK-12553: function JACK_HOOK throws a Bus error and crashes Asterisk |
ASTERISK-12554: JACK_HOOK will occasionally stream extremely high values that sound like static. |
ASTERISK-12555: Call on Queue do not are delivery to free agents |
ASTERISK-12556: Impossible to retrieve privacy info in a PRI incoming call |
ASTERISK-12557: [patch] app_queue on misdn channels plays wrong MoH |
ASTERISK-12558: CDR TDS does notbreport the userfield contents,as cdr_odbc does |
ASTERISK-12559: Distorted audio |
ASTERISK-12560: [patch] Coding Guidelines in TeX format. |
ASTERISK-12561: New application to indicate if a call is encrypted |
ASTERISK-12562: New config parameter to enforce encryption |
ASTERISK-12563: [patch] Missing UniqueId header in ParkedCallGiveUp and ParkedCallTimeOut manager events |
ASTERISK-12564: 200 OK retransmitted even when first SIP OK is ACKed |
ASTERISK-12565: Mixmonitor crashes asterisk - problem in translation path? |
ASTERISK-12566: Changes to NoCDR() prevent access to certain CDR variables post call hangup. |
ASTERISK-12567: The field CDR(userfield) does not get written to the database |
ASTERISK-12568: Not able to make outgoing call [Failed to authenticate on INV ITE] |
ASTERISK-12569: ASTERISK FAX2MAIL |
ASTERISK-12570: We could NOT get the channel lock for SIP |
ASTERISK-12571: crash after "Hard hangup called while fd is blocked" |
ASTERISK-12572: [patch] Dynamic realtime capability for FindMe/FollowMe |
ASTERISK-12573: "From" shouldn't be matched against "users" if INVITE arrives from a "peer" IP |
ASTERISK-12574: [patch] Replace call to deprecated function |
ASTERISK-12575: [patch] Fix debug thread locals compilation |
ASTERISK-12576: [patch] asterisk crashes when SPRINTF function has too few arguments |
ASTERISK-12577: Asterisk crashes, leaving a voicemail message stored on pgsql DB via ODBC |
ASTERISK-12578: [patch] AMI UpdateConfig -- When creating category, memory allocation failure in config.c |
ASTERISK-12579: Unable to hangup channel SIP to Voicemail |
ASTERISK-12580: [patch] Memory leak while trying to free a not existent or moved pointer. |
ASTERISK-12581: AEL while loop - high cpu usage and crash |
ASTERISK-12582: [patch] Optionally use black on white for the terminal settings |
ASTERISK-12583: [patch] Asterisk 1.4 (all versions) does not work on Mac OS X -- poll.c |
ASTERISK-12584: ztdummy gives unresolved symbols on Linux kernel 2.6.26.1 |
ASTERISK-12585: [patch] chan_sip does not always create regexten for registering peers |
ASTERISK-12586: [patch] Memory leak in chan_gtalk while trying to unload the module. |
ASTERISK-12587: [patch] Memory leak while unloading the module. |
ASTERISK-12588: crash when subscribe to pattern match for a hint |
ASTERISK-12589: Include statements in extensions.conf.sample |
ASTERISK-12590: Dial h option not operating as defined when disconnect redefined in features.conf |
ASTERISK-12591: Realtime registrations don't work after a sip reload |
ASTERISK-12592: voice file updation issue |
ASTERISK-12593: [patch] Missing colon in To/From headers of RTCP manager events |
ASTERISK-12594: voice file updation issue |
ASTERISK-12595: SayDigits |
ASTERISK-12596: not reading new sip peers/users |
ASTERISK-12597: [patch] Missing UniqueId header in ParkedCall manager events |
ASTERISK-12598: [patch] Missing \r\n in JitterBufStats manager event |
ASTERISK-12599: No new line between parameters |
ASTERISK-12600: [patch] Column width with cli command sip/iax2 show registry |
ASTERISK-12601: pattern match for a hints always gives state:idle for all extensions |
ASTERISK-12602: Configure does not find the tds libraries |
ASTERISK-12603: Outgoing Proxy on SIP Register |
ASTERISK-12604: [patch] DISA does not accept extensions beginning with "#" |
ASTERISK-12605: [patch] chan_sip is leaking in build_peer |
ASTERISK-12606: mysql cdr not loging custom fields |
ASTERISK-12607: [patch] Response events to CoreShowChannel are missing the Event header |
ASTERISK-12608: Multiple agents answering a call. |
ASTERISK-12609: [patch] fix mohinterpret and mohsuggest settings from general section in chan_iax2.c |
ASTERISK-12610: wrong SRV query |
ASTERISK-12611: Crash in chanspy mutex |
ASTERISK-12612: AEL parser should trim RHS of an assignment |
ASTERISK-12613: [patch] handle_getvariable doesn't initialize workspace |
ASTERISK-12614: [patch] Problem using UpdateConfig AMI command when dstfilename points to non-existing file |
ASTERISK-12615: [patch] Missing doc for SipShowRegistry action and RegistryEntry event |
ASTERISK-12616: Local channel does not support exten/callerid style dialplan entries (ast_exists_extension placement). |
ASTERISK-12617: [patch] Adds "skinny show debug" that provides device, line and sub info for debugging purposes. |
ASTERISK-12618: Deadlock in sip_alloc |
ASTERISK-12619: [patch] Response to SipShowPeer manager action is malformed |
ASTERISK-12620: [patch] 'core show sysinfo' on systems that dont have sysinfo but do have sysctl |
ASTERISK-12621: [patch] Manager action SipNotify disconnecting the manager session on error |
ASTERISK-12622: out stream is not recorded and only in stream getting recorded. |
ASTERISK-12623: [patch] check correct tags during REFER in pedantic mode |
ASTERISK-12624: IAX variable feature does not work |
ASTERISK-12625: Dial(SIP/exten@host:port) ignores port |
ASTERISK-12626: No Account Code Set For Blind Transfers |
ASTERISK-12627: AEL doesn't like using dialplan function call as argument for 'jump' statement |
ASTERISK-12628: [patch] Incorrect checking of the "Channel" header on Originate action |
ASTERISK-12629: how to disallow the native bridge between the two channels |
ASTERISK-12630: ringall and queues.conf timeout don't work if queue members are agents |
ASTERISK-12631: [patch] Fix for trace frames compile option |
ASTERISK-12632: [patch] Device Side transfer of a call between 2 extensions leads to failure because MACRO_DEPTH is not reset |
ASTERISK-12633: CDRs produced on blind transfer are incorrect |
ASTERISK-12634: Asterisk fails to start |
ASTERISK-12635: CDR billsec value accuracy |
ASTERISK-12636: can any one help 4 this my card is TDM2400P digium |
ASTERISK-12637: [patch] chan_local doesn't copy the dialplan (cid.cid_ton) into the new channel |
ASTERISK-12638: [patch] Add variables property to AgentRingNoAnswer manager event |
ASTERISK-12639: prompt tokens defined in followme.conf within followmeid prompt tokens are inconsistent |
ASTERISK-12640: [patch] Outgoing notification hints not functioning for SIP type=friend |
ASTERISK-12641: Playback does not play videofiles |
ASTERISK-12642: Asterisk 1.6.0-beta9 doesn't compile under Open Solaris |
ASTERISK-12643: Aaterisk doesn't build on OpenSolaris (SVN) |
ASTERISK-12644: Unable to access mailbox options when you have a temporary greeting |
ASTERISK-12645: chanspy with DTNF detect |
ASTERISK-12646: can any one help 4 this my card is TDM2400P digium |
ASTERISK-12647: [patch] Mysql CDR logger should not DESC the table every time through. |
ASTERISK-12648: [patch] pthread_cancel segmentation faults |
ASTERISK-12649: Wrong branch on CANCEL after SIP INFO in early dialog |
ASTERISK-12650: [patch] Turn off qualify on uncached realtime peers |
ASTERISK-12651: Asterisk doesn't respect the codec order |
ASTERISK-12652: [patch] Reason header support |
ASTERISK-12653: [patch] Prompt for immediate removal of modules not installed by current version of Asterisk |
ASTERISK-12654: sip registeration with ldap |
ASTERISK-12655: Parking is happening only one time |
ASTERISK-12656: Call failed to go through, reason (8) Congestion (circuits busy) |
ASTERISK-12657: SIP channel never gets destroyed - Tx: BYE - response 404 |
ASTERISK-12658: [patch] Attended transfers do not call update_queue until after transfered call ends |
ASTERISK-12659: Not able to put call on hold |
ASTERISK-12660: Asterisk 1.6.0beta9 Crashed with segmentation fault |
ASTERISK-12661: [patch] Channel name buffer is too small |
ASTERISK-12662: [patch] POSIX thread operations errors |
ASTERISK-12663: Transport type not correctly displayed in "sip show peer" output |
ASTERISK-12664: Read() does not set variable |
ASTERISK-12665: Multiple hangups on IAX trunk between two asterisk servers |
ASTERISK-12666: [patch] Commands issued to asterisk using a remote console on OSX have no effect |
ASTERISK-12667: [patch] T38 gateway |
ASTERISK-12668: Need to generic basic dial string.... |
ASTERISK-12669: AEL switches inside macros not working like extensions.conf's ones |
ASTERISK-12670: [patch] send rel with unallocated cause code insted of normal call clearing when call invalid extension |
ASTERISK-12671: [patch] Huge memory leak because memory of channel cdr struct is never returned |
ASTERISK-12672: crash in chan_iax |
ASTERISK-12673: [patch] undesired autoprune behavior might delete all your contacts |
ASTERISK-12674: [patch] Rewrite the config stuff to seperate lines and devices |
ASTERISK-12675: After retrieved the parked call from SIP phone unable to bilnd transfer or atttend transfer |
ASTERISK-12676: Mailbox greetings not working w/IMAP |
ASTERISK-12677: Centos5 can't work SRTP |
ASTERISK-12678: Function CUT doesn't work if passed as parameter to macro in AEL |
ASTERISK-12679: Asterisk stopped compiling on Open Solaris [revision 140974] |
ASTERISK-12680: '*' escape from VoiceMail no longer works |
ASTERISK-12681: Can't get lock in sip to sip calls |
ASTERISK-12682: [patch] ss7 dual seizure |
ASTERISK-12683: Generating output without additional NoOp() |
ASTERISK-12684: r140417 broke sip nat interoperability |
ASTERISK-12685: One-touch parking results in stuck/deadlocked channel if parked channel hangs up while announcement still being played |
ASTERISK-12686: [patch] OpenH323 detection problem |
ASTERISK-12687: Crash in ast_cdr_start() when Local channel is involved |
ASTERISK-12688: [patch] outboundproxy=proxy.mmmydomain.net where domain can not be resolved silently removes the sip section |
ASTERISK-12689: [patch] 'core show sysinfo' on systems without HAVE_SYSINFO but with HAVE_SYSCTL |
ASTERISK-12690: Revision 141028 (to resolve issue #11979) breaks compatibility with some agi scripts |
ASTERISK-12691: Asterisk 1.4.21.2 crash ast_do_masquerade (segfault at 00000000000000d8 rip 0000003ad54082f9 rsp 0000000040523fb0 error) |
ASTERISK-12692: Call disconnection with branch in via header changing and scheduled destruction of sip call |
ASTERISK-12693: No TO tag after SIP INFO in early dialog |
ASTERISK-12694: 'sip show peers' shows peers in UNKNOWN state after chan_sip reload |
ASTERISK-12695: wait_for_answer never receives HANGUP frame sent via ast_queue_hangup |
ASTERISK-12696: rc5: chan_dahdi doesn't recognize TDM400P channels |
ASTERISK-12697: After update from 140415 to 141991 get crash in few minutes after start |
ASTERISK-12698: add user in config how one line |
ASTERISK-12699: [patch] Allows Action: Register in Manager API to allow asterisk to register to a remote server |
ASTERISK-12700: Multiple Hints do not work correctly |
ASTERISK-12701: Queues with Dynamic members like "Local/" can't get status of member and sometime get second call from queue |
ASTERISK-12702: In SIP to PSTN call, CDR disposition is ANSWERED always |
ASTERISK-12703: [patch] Audible clicks when playing a file with header larger than AU_HEADER_SIZE |
ASTERISK-12704: func odbc readsql is broken in Trunk |
ASTERISK-12705: skinny memory leak |
ASTERISK-12706: 0007068: Asterisk crashes when using record_file to record in h263 format |
ASTERISK-12707: Need to remove "silenceSupp" parameter from SDP |
ASTERISK-12708: Make asterisk push an event to the manager in the case of a transfer. |
ASTERISK-12709: [patch] Timestamp in DTMF does not match the one in the voice stream |
ASTERISK-12710: Crash - call to agent channel when agent is logging in. |
ASTERISK-12711: H and h parameters do not hang up when * is dialed |
ASTERISK-12712: CLI> "file convert file.ogg file.gsm" Segfaults |
ASTERISK-12713: Call from a Polycom, trough Asterisk, to a Local/ + DAHDI analog channel. Hangup DAHDI and crash. |
ASTERISK-12714: asterisk console screws up terminal subtly when exited with ctrl-c with some shells |
ASTERISK-12715: R141503 reverted fix for one-touch parking crash and asterisk crashes again |
ASTERISK-12716: Unable to run menuselect |
ASTERISK-12717: Seg fault 1.6.0 trunk |
ASTERISK-12718: IAX Transfer/releasing between 3 asterisk's are not working. |
ASTERISK-12719: No pseudo device with DAHDI |
ASTERISK-12720: Func CUT requires default of '-' to be entered |
ASTERISK-12721: exten => s,n,SMS(sms,2a) causes seg fault |
ASTERISK-12722: Make Menuselect fails |
ASTERISK-12723: [patch] Asterisk 1.6.0-rc6 crashes with ReceiveFAX |
ASTERISK-12724: [patch] usereqphone parameter doesn't work |
ASTERISK-12725: [patch] add CLI command 'agents2 show available' in app_agents.c |
ASTERISK-12726: 5277 Segmentation fault |
ASTERISK-12727: rc6: no dial tone on chan_dahdi |
ASTERISK-12728: [patch] Only one custom feature can be executed simultaneously |
ASTERISK-12729: menuselect in asterisk-addons-1.6.0-rc1 |
ASTERISK-12730: [patch] remove zap/dahdi-related code duplications with some #define-s |
ASTERISK-12731: Agent state not updating using both dynamic and direct agents. |
ASTERISK-12732: After installing PBX, unable to make SIP to PSTN calls most of the times |
ASTERISK-12733: [patch] ${VM_CALLERID} string content varies between email subject and body when callidname is NULL |
ASTERISK-12734: [patch] mISDN rejects incoming calls |
ASTERISK-12735: Asterisk crash 1.4 SVN |
ASTERISK-12736: crash in ast_cdr_start, backtraces attached |
ASTERISK-12737: unable to place outgoing call on TE Port |
ASTERISK-12738: Configure res_snmp failure |
ASTERISK-12739: In some cases parking extension isn't removed after parked channel is gone (continuation of problem reported in #13425) |
ASTERISK-12740: One-touch parking failure results in the call drop, while parties should be able to continue conversation |
ASTERISK-12741: [patch] Crash in res_musiconhold |
ASTERISK-12742: freeing unused memory in ast_get_enum |
ASTERISK-12743: 0007016: Nonce blanked in Elastix system |
ASTERISK-12744: Cannot send more than one variable to a python script |
ASTERISK-12745: Arbitrarily set the release cause on an inbound h.323 channel |
ASTERISK-12746: Memory leak in channel variables |
ASTERISK-12747: Revision 142675 broke one-touch parking initiated by caller. |
ASTERISK-12748: Crash in ast_cdr_start() during one-touch parking if parking extension set with PARKINGEXTEN already busy |
ASTERISK-12749: Loss of incoming audio during a phone call through a SIP trunk |
ASTERISK-12750: acf_channel_read doesn't check if p is null |
ASTERISK-12751: Instability with V1.4.X |
ASTERISK-12752: Recording transcoded call with Monitor() uses 3 licenses of G.729 |
ASTERISK-12753: MYSQL INSERT Query length |
ASTERISK-12754: zap callerid in cdr not coming for incoming calls |
ASTERISK-12755: [patch] Incorrect calculation of Realtime conference announcements |
ASTERISK-12756: Trouble with Temporarily Moved |
ASTERISK-12757: [patch] Fixes skinny unload |
ASTERISK-12758: hint change state to Idle when peer reregisters |
ASTERISK-12759: Last 1.4 svn crash. Backtrace attached |
ASTERISK-12760: Audio passed through during dial macro. |
ASTERISK-12761: [patch] Hold logic broken? |
ASTERISK-12762: dundi lookups occasionally stop working |
ASTERISK-12763: video_src_res and video_dest_res are write-only in ast_rtp_early_bridge |
ASTERISK-12764: [patch] Recording stops after Transfer when using MixMonitor() |
ASTERISK-12765: Memory leacks in stress test with failure |
ASTERISK-12766: Channel re-invited on destination ringing not re-invited back if ringing abandoned. |
ASTERISK-12767: Partial writes on Manager API |
ASTERISK-12768: [patch] Asterisk crash getting fax by sip channel |
ASTERISK-12769: exten = 2813,n, Queue(test,c,,,,,,inqueue) craches when went into invalid extension |
ASTERISK-12770: setinterfacevar is missing from queue_table in contrib/scripts/realtime_pgsql.sql |
ASTERISK-12771: asterisk died after transfer |
ASTERISK-12772: [patch] skinny support for res_phoneprov |
ASTERISK-12773: Mixmonitor doens't record call after attended transfer (atxfer) |
ASTERISK-12774: Joining a MeetMe conference and hanging up shortly after results in SIGSEGV |
ASTERISK-12775: WARNING: Out of buffer space (IAX-Trunk and speex codec ) |
ASTERISK-12776: [patch] Asterisk won't compile against uclibc |
ASTERISK-12777: [patch] pbx_lua does not allow numeric strings as extension numbers |
ASTERISK-12778: [patch] pbx lua leaks memory after application error |
ASTERISK-12779: [patch] Background and WaitExten broken in pbx_lua |
ASTERISK-12780: [patch] add a traceback to pbx_lua execution errors |
ASTERISK-12781: Calls originated from AMI do not have channel variables specified in sip.conf set |
ASTERISK-12782: ast_moh_free_class in res_musiconhold.c:195 |
ASTERISK-12783: Segment Fault in asterisk "pthread_cancel" |
ASTERISK-12784: An application to do SLINEAR recording and carrier detection |
ASTERISK-12785: Asterisk sending the wrong codec on re-invite. |
ASTERISK-12786: [patch] Malformed registration line is copied verbatim in To and From headers |
ASTERISK-12787: [patch] asterisk -rx writes history |
ASTERISK-12788: WaitForSilence() sometimes doesn't always wait when using SIP and a callfile |
ASTERISK-12789: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions |
ASTERISK-12790: [Patch] Fix false string allocation in realtime_pgsql_store where creating INSERT |
ASTERISK-12791: app_rpt does not compile against DAHDI in Asterisk 1.4 |
ASTERISK-12792: [feature request] Reporter would like an option to ignore src ports in iax2 |
ASTERISK-12793: Bang not showing up in <tab> on CLI |
ASTERISK-12794: blindxfer doesn't work properly |
ASTERISK-12795: ignoreregexpire does not work as expected |
ASTERISK-12796: [patch] AGI processes going to defunct |
ASTERISK-12797: Asterisk wont compile if gmime-devel lib is not installed |
ASTERISK-12798: blindxfer doesn't work! |
ASTERISK-12799: crash with res_features |
ASTERISK-12800: meetme list <confno> concise fails |
ASTERISK-12801: [patch] Make 'core show functions like xxxxx' case insensitive. |
ASTERISK-12802: Wav49 corrupted in voicemail |
ASTERISK-12803: STRFTIME returns incorrect time |
ASTERISK-12804: [patch] MALLOC_DEBUG causes crash in chan_h323 |
ASTERISK-12805: SMS help is incorrect, Typo |
ASTERISK-12806: SMS receive file name incorrect |
ASTERISK-12807: please see 0013468 |
ASTERISK-12808: clid will only set number not name |
ASTERISK-12809: Occassional failure to connect pickup call with park call |
ASTERISK-12810: [patch] When unregistering a UA, 200 OK response from Asterisk is not SIP compliant |
ASTERISK-12811: [patch] Crash in decode_length - udptl.c:159 |
ASTERISK-12812: Add reboot-snom to sip_notify |
ASTERISK-12813: [patch] Bad handling of Contact header, which should not be present in 1XX responses to REGISTER, but also in several other case |
ASTERISK-12814: [patch] Database from extconfig.conf ignored |
ASTERISK-12815: error: ‘PTHREAD_MUTEX_RECURSIVE’ undeclared (first use in this function) on powerpc during compile of 1.6 rc6 |
ASTERISK-12816: [patch] Compile error with IMAP_STORAGE due to removed autoconfig.h in Makefile |
ASTERISK-12817: variables set in sip.conf using "setvar=VARIABLE_NAME=xxxx" are not honoured when doing a blind transfer |
ASTERISK-12818: autopause=no not working with Asterisk 1.4.22rc5 |
ASTERISK-12819: Asterisk > 1.4.17 not mixing recorded channels calls with soxmix on Debian Etch |
ASTERISK-12820: no warning message returned when soxmix is not installed |
ASTERISK-12821: after upgrading to asterisk -1.4.22 zaptel features are shown as dahdi features |
ASTERISK-12822: [patch] SMS app ignores parameter 'p' - initial pause |
ASTERISK-12823: Asterisk 1.6.0 will not compile on OS X 10.5.5 |
ASTERISK-12824: sip debug lower levels |
ASTERISK-12825: Asterisk 1.4.15 random crashes |
ASTERISK-12826: Asterisk segfaults when using SIP session timers |
ASTERISK-12827: When calling party hangup the line followme app continue ringing. |
ASTERISK-12828: Dial with timeout 0 places a call and immediately cancels it. |
ASTERISK-12829: [patch] CANCEL before Trying |
ASTERISK-12830: IAX port change using dnsmgr |
ASTERISK-12831: IAX2 peer does not show trunk |
ASTERISK-12832: [patch] Potential spiral detected problem |
ASTERISK-12833: [patch] Fields have a maximum length of 255 chars |
ASTERISK-12834: [patch] Shortcut for duplicating the last extension, so the pattern doesn't need to be repeated |
ASTERISK-12835: Asterisk won't register if SIP-port at peer differ than local (5060) |
ASTERISK-12836: Asterisk fills "via" header not correctly. |
ASTERISK-12837: SRV record failures on OpenBSD (all OSes?) |
ASTERISK-12838: [patch] Remove DAHDI operator mode special casing in app_dial |
ASTERISK-12839: Missing userfield for Queue call with NO ANSWER |
ASTERISK-12840: Audio not passing between two Asterisk boxes when OpenSER in the middle |
ASTERISK-12841: [patch] Make format_ogg_vorbis work on OpenBSD |
ASTERISK-12842: [patch] Crash in cdr code in specific one-touch parking scenario |
ASTERISK-12843: OpenBSD 4.2 launch failures |
ASTERISK-12844: [patch] Messages not deleted properly when delete=yes in voicemail.conf |
ASTERISK-12845: [patch] VM_CALLERID yields different results if CID is null or empty |
ASTERISK-12846: insecure doesn't work |
ASTERISK-12847: chan_iax2 isn't using HANGUP anymore? |
ASTERISK-12848: [patch] ignores 'p' parameter, received filename wrong, protocol 2 fixed, typo in help, remove extra braces |
ASTERISK-12849: unable to change sip from message from "Unknown" to anything else using set(callerid(num)=xxxx) |
ASTERISK-12850: "No entry in voicemail config file" error for all RealTime VoiceMail Users |
ASTERISK-12851: compilation fail on FreeBSD |
ASTERISK-12852: Change in behavior for Zap dialing |
ASTERISK-12853: [patch] Make func_realtime work more like app_realtime |
ASTERISK-12854: [patch] Asterisk IMAP headers are not processed correctly |
ASTERISK-12855: [patch] Shared IMAP mailboxes can cause the server to crash |
ASTERISK-12856: extensions.conf refers to two "exten => formats", but the second one has been deleted from the text |
ASTERISK-12857: Jabber fails to authenticate when using SSL. |
ASTERISK-12858: [patch] 16kHz wav format |
ASTERISK-12859: WAITSTATUS will never get set for digitally muted channels |
ASTERISK-12860: [patch] User not notified that a temporary greeting is active when using ODBC voicemail |
ASTERISK-12861: Provide working demo users for ldap/realtime backend. |
ASTERISK-12862: voice call disconnect |
ASTERISK-12863: Asterisk SIP calls stop working having more than 300 calls (more than 600 channels) |
ASTERISK-12864: Dundi and DNS |
ASTERISK-12865: deleting and moving voicemessages around |
ASTERISK-12866: Wrong CDR is posted if call files or Manager API Originate is used |
ASTERISK-12867: [patch] CLI command 'agi show commands [topic]' is not working as expected. |
ASTERISK-12868: [patch] Asterisk init script for archlinux and patch for MakeFile |
ASTERISK-12869: sip show inuse count is negative |
ASTERISK-12870: If the user hangup during recording, recorded file isn't removed |
ASTERISK-12871: No audio when dialling number that has Telco announcement "area code 626 has changed to ..... " |
ASTERISK-12872: [patch] make_email_file() uses different values for "cidname" and "enc_cidname" (0013643 issue unresolved) |
ASTERISK-12873: asterisk random crashes: IMAP toolkit crash: Unlock when not locked |
ASTERISK-12874: Additional codecs are added to the SDP after a "Moved Temporarily" mesage - SIP TCP |
ASTERISK-12875: [patch] Addition of a Mailbox id facility to allow shared mailboxes to work |
ASTERISK-12876: [patch] app_sms doesn't answer the call, currently requires Answer() before hand |
ASTERISK-12877: channel get stuck on ast_queue_frame when hanging up |
ASTERISK-12878: [patch] make code conform to trunk format guidelines |
ASTERISK-12879: tcptls.c: ast_make_file_from_fd() memory leak if DEBUG_THREADLOCALS defined. |
ASTERISK-12880: Backport astConfigCallsActive and astConfigCallsProcessed MIB objects to Asterisk 1.4 |
ASTERISK-12881: Crash on peer notify |
ASTERISK-12882: [patch] Asterisk sleeps forever in poll() when terminating both SIP endpoints of a bridged channel |
ASTERISK-12883: Segmentation Fault with 1.4.21.2 in rtp.c:1131, Application chanSpy |
ASTERISK-12884: SegFault when after saying "Thank you" when user presses # or just hangs up |
ASTERISK-12885: [patch] Console/dsp not hanging up after playing sound file. |
ASTERISK-12886: Documentation about application AGI |
ASTERISK-12887: [patch] Update app_fax to work with spandsp-0.0.6 |
ASTERISK-12888: Asterisk 1.4.21.2 losing all ability to make calls |
ASTERISK-12889: CallerID not sent to SIP stations in SLA |
ASTERISK-12890: [patch] Unanswered Queue() calls don't have CDR |
ASTERISK-12891: [patch] atxfer from PSTN to SIP exten gives "Unexpected control subclass '-1'", legs cannot hear each other |
ASTERISK-12892: [patch] AST-2009-001 |
ASTERISK-12893: I get two MWI subscritions for a peer (phone) |
ASTERISK-12894: [patch] Comply with trunk coding guidlines |
ASTERISK-12895: [patch] Comply with trunk coding guidlines |
ASTERISK-12896: [patch] Support a 'no-patterns' option for the Realtime switch |
ASTERISK-12897: [patch] [branch] appdocsxml conversions for help |
ASTERISK-12898: [patch] peer outboundproxy-ptr is copied to dialog and then freed |
ASTERISK-12899: [patch] Memory leak while trying to free a malloced memory with an ast_free() call instead of just free(). |
ASTERISK-12900: Abort in free in queue show <name> command |
ASTERISK-12901: Asterisk has new problems compiling on Solaris |
ASTERISK-12902: Can't use type=friend anymore in sip.conf |
ASTERISK-12903: [patch] SIP_NOTIFY message incorrectly builds address in the To: tag |
ASTERISK-12904: Queue did not take care of h option |
ASTERISK-12905: [patch] fix content of CHANGES file for agents.conf |
ASTERISK-12906: [patch] fix content of agents.conf.sample |
ASTERISK-12907: [patch] MWI NOTIFY always tries to use UDP, even if the peer is connected via TCP |
ASTERISK-12908: AGI-program receives SIGHUP on hangup although AGISIGHUP is set to "no" |
ASTERISK-12909: Hangups are not recognized when using IAX2 encryption |
ASTERISK-12910: [patch] Inband DTMF on outbound call is not detected when dtmfmode=auto |
ASTERISK-12911: [patch] invites with proxy_auth have wrong via-branch-tag |
ASTERISK-12912: [patch] Using SIP_HEADER in AMI with NULL channel causes crash |
ASTERISK-12913: [patch] Prevent false answer of channel when going off-hook |
ASTERISK-12914: [patch] 1.6.0.1 crashes randomly |
ASTERISK-12915: [patch] Neither CHANGES, nor UPGRADE.txt for 1.6.0 say quotes are no longer stripped |
ASTERISK-12916: Unable to call to the "no conventional" numbers. |
ASTERISK-12917: [patch] Replaced reload by module reload in contrib/init.d |
ASTERISK-12918: [patch] res_ais won't build on 64bit |
ASTERISK-12919: Answered calls have no duration on Asterisk 1.4.22 |
ASTERISK-12920: current trunk chan_sip does not compile in devmode |
ASTERISK-12921: [patch] ERROR[7387]: res_config_ldap.c:1292 update_ldap: Couldn't modify dn:cn=1001,dc=xxx,dc=xxx because Invalid syntax |
ASTERISK-12922: [patch] chan_dahdi.c:3693: error: struct zt_params has no member named chan_alarms |
ASTERISK-12923: [patch] resolving hostnames should ignore uri-paramters |
ASTERISK-12924: After upgrading to 1.4.22 from 1.4.21.1 CDRs behave wrongly |
ASTERISK-12925: Asterisk randomly crashes when using chan_misdn |
ASTERISK-12926: [patch] CLI command 'channel request hangup <channel>' not working as expected. |
ASTERISK-12927: [patch] missing channel's uniqueid field in AMI's queue status |
ASTERISK-12928: IAX2 qualify problems between 1.6 and 1.4 boxes |
ASTERISK-12929: [patch] Add ability to register callback that should run just prior to when asterisk is fully loaded |
ASTERISK-12930: Indications are not passed from old peer to new peer during masquerade |
ASTERISK-12931: [patch] Allow to override the default prompt (agent-pass) on Authenticate application |
ASTERISK-12932: [branch] Add CLI aliases module to asterisk |
ASTERISK-12933: [patch] Answer() doesn't when the state is Dialing |
ASTERISK-12934: [patch] Overlap dialing blocks |
ASTERISK-12935: [patch] Event processing sometimes hangs when using res_timing_pthread |
ASTERISK-12936: Compile fails at asterisk.c |
ASTERISK-12937: Seg. fault when including missing extensions file |
ASTERISK-12938: Blackberry Pearl (8100) rarely initialises |
ASTERISK-12939: Recordings out of sync when using chanspy |
ASTERISK-12940: IAX2 storm (type 4, subtype 20: AST_CONTROL_SRCUPDATE) |
ASTERISK-12941: All Call Recordings are world readable [Security Risk] |
ASTERISK-12942: Dial with M(macro) doesn't work with AEL macros |
ASTERISK-12943: [patch] Set a sane umask inside safe_asterisk |
ASTERISK-12944: dahdi fails compile on powerpc |
ASTERISK-12945: Avoiding deadlock for channel problem |
ASTERISK-12946: app_fax failing to build on powerpc / spandsp6 / asterisk1.6.0.1 |
ASTERISK-12947: Moved Temporarily is unable to create local channel and fails |
ASTERISK-12948: [patch] Obvious typo (logic error) |
ASTERISK-12949: [patch] Prevent a crash when unloading res_musiconhold.so and then stopping asterisk. |
ASTERISK-12950: 302 Redirect (forward no answer) to bad extension causes channel to be left up (Ringing) |
ASTERISK-12951: after return from child context ARG is cleared |
ASTERISK-12952: [patch] On Mac OS X PowerPC, Asterisk 1.6.0.1 cannot create outbound channels |
ASTERISK-12953: voicemail indicator does not clear when last message is read or deleted via IMAP |
ASTERISK-12954: Cannot record sounds in WAV and WAV49 longer then 5-8 seconds |
ASTERISK-12955: Asteris never responce ACK for 487 Request Terminated |
ASTERISK-12956: asterisk-addons-1.6-current.tar.gz is missing |
ASTERISK-12957: Default delimiter for ${CUT(var,,n)} is no longer handled |
ASTERISK-12958: [patch] New option to specify mysql charset |
ASTERISK-12959: Dundi and DNS |
ASTERISK-12960: crash or dialing isn't possible |
ASTERISK-12961: asterisk blocked at startup between main/asterisk.c/loader.c/load_modules and manager.c/loader.c/ast_module_reload |
ASTERISK-12962: sip show peer doesnt working |
ASTERISK-12963: Return to voicemail after dial-out |
ASTERISK-12964: dahdi doesn't send correct dtmf on dialing |
ASTERISK-12965: [patch] Cannot register with sip providers that require '@' in the username |
ASTERISK-12966: segfault in res_odbc.c when rtupdate=yes in sip.conf with MS-SQL ODBC |
ASTERISK-12967: [patch] Memory leak while reloading chan_mgcp.so |
ASTERISK-12968: [patch] DAHDI_CHECK_HOOKSTATE automatically defined when chan_dahdi is built with zaptel support |
ASTERISK-12969: KEYPADHASH returns incorrect values |
ASTERISK-12970: [patch] mISDN broken for BRI always reporting all chanells in use |
ASTERISK-12971: [patch] Application not accept any option after deleting jump+101 |
ASTERISK-12972: Queue stats are no longer reset by 'reload' or 'module reload app_queue.so' |
ASTERISK-12973: app_dial doesn't report back DIALSTATUS, ANSWEREDTIME and DIALEDTIME |
ASTERISK-12974: [patch] CDR for picked up parked call gives answer time < start time and no record for parking |
ASTERISK-12975: [patch] Incorrect use of sizeof() |
ASTERISK-12976: Realtime queueing using AddQueueMember |
ASTERISK-12977: [patch] forkcdr() doesn't fork when call disposition is ANSWERED |
ASTERISK-12978: Drop outbound call to IVR in early media. |
ASTERISK-12979: stdexten no longer logs incoming dialled digits in CDR |
ASTERISK-12980: crash after transfer |
ASTERISK-12981: [patch] No way to tune talker optimization in meetme, causes users to get cut off while they're still talking |
ASTERISK-12982: manager core dumps after 45,000 AMI originates |
ASTERISK-12983: After receiving a 302 Temporarily Moved response, call is not sent in to users defined context |
ASTERISK-12984: app_directory crashses Asterisk when voicemail entry doesn't have a name |
ASTERISK-12985: Adding a channel to a SPYGROUP when there is already a listener causes segfault |
ASTERISK-12986: codecs settings does work only in device specific section |
ASTERISK-12987: [patch] Missing mutex unlock on error inside local_call(). |
ASTERISK-12988: Asterisk restarted automatically |
ASTERISK-12989: insufficent log information |
ASTERISK-12990: [patch] Integer divide by zero |
ASTERISK-12991: peer matching issue when use type=peer |
ASTERISK-12992: chan_gtalk and app_jack don't work together when originating a call from the CLI |
ASTERISK-12993: Music On Hold stops sending outbound RTP after first inbound RTP packet is lost |
ASTERISK-12994: Missing CallerID in CDR when last app is voicemail |
ASTERISK-12995: [patch] Do not expose ast_str internals. |
ASTERISK-12996: not recording the calls |
ASTERISK-12997: MixMonitor by features crash |
ASTERISK-12998: [patch] clearing expired entries from /dundi/cache |
ASTERISK-12999: Executing 'h' extension if parked channel hangs up |